Good day, everyone!
Then make call to SIP, i see INVITE message.
For example:
Summary
Session Initiation Protocol (SIP as raw text)
INVITE sip:45077@10.15.81.30 SIP/2.0\r\n
Via: SIP/2.0/TCP 10.1.96.65:39995;branch=z9hG4bKyy3U73Sy0aygN\r\n
Max-Forwards: 70\r\n
From: sip:77772257247@10.15.81.30;tag=y968t478j6tQj\r\n
To: sip:45077@10.15.81.30\r\n
Call-ID: ObPgw3yUySsdES9tuC9x5AG\r\n
CSeq: 983001120 INVITE\r\n
Contact: sip:77772257247@10.1.96.65:39995;transport=tcp\r\n
User-Agent: Janus WebRTC Server SIP Plugin 0.0.9\r\n
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, REFER, MESSAGE, INFO, NOTIFY\r\n
Supported: replaces\r\n
Content-Type: application/sdp\r\n
Content-Disposition: session\r\n
Content-Length: 795\r\n
\r\n
v=0\r\n
o=- 254732666825406433 4258817811759427564 IN IP4 10.1.96.65\r\n
s=-\r\n
t=0 0\r\n
m=audio 39538 RTP/AVP 111 63 9 0 8 13 110 126\r\n
c=IN IP4 10.1.96.65\r\n
a=rtpmap:111 opus/48000/2\r\n
a=fmtp:111 minptime=10;useinbandfec=1\r\n
a=rtpmap:63 red/48000/2\r\n
a=fmtp:63 111/111\r\n
a=rtpmap:9 G722/8000\r\n
a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:8 PCMA/8000\r\n
a=rtpmap:13 CN/8000\r\n
a=rtpmap:110 telephone-event/48000\r\n
a=rtpmap:126 telephone-event/8000\r\n
a=mid:0\r\n
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\n
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\n
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\n
a=msid:7a7b60f2-6bf2-42ab-9150-93ae4b9f42f5 db27268b-2352-4b9c-8a42-e2d8c17fca93\r\n
a=rtcp-fb:111 transport-cc\r\n
And my question about port in “Via:”
Via: SIP/2.0/TCP 10.1.96.65:39995;branch=z9hG4bKyy3U73Sy0aygN\r\n
Is there a way to control the range from which a given port is selected?
Best Regards, Vitaliy.